It has been possible for some time to exploit the inexpensive data communications that are available via Internet Protocol networks to make very low cost international telephone calls and domestic telephone calls. Users communicate using their computers, which transform voice signals into packets of compressed digital audio information that are then sent over the Internet Protocol network. Part of the cost saving for wireline networks results from using a single network for sending voice and data, and in particular the ability to exploit the spare capacity of the available networks. The transfer of voice signals using an Internet Protocol network is referred to as ‘voice over IP’ or ‘VoIP’.
A program that has contributed to the success of VoIP and which enables managing VoIP calls from a computer is the Skype™ program and service. This allows users to make free voice calls to other Skype users anywhere in the world, or to call fixed-line telephones or mobile telephones for a fee. The Skype program includes support for instant messaging, file transfer, short message service (SMS) and video conferencing and so provides access from a computer to multiple communication channels.
Although this ability to transfer voice signals using an internet protocol network can represent very good value to users who make regular international calls, as well as being usable for domestic calls and other data services, its adoption has not been as widespread as might be expected. Many people prefer to continue making voice calls via their fixed-line telephones or mobile telephones even when these calls are more expensive. The reason for this limited adoption is partly the proactive effort required by users to configure a computer to make use of VoIP services. For example, a user typically has to acquire and install new software on their home computer, whereas typical mobile telephones have been unable to provide VoIP capability. Another reason for low adoption of VoIP services is the constraint of having to be in close proximity to the computer that provides the VoIP service.
Mobile client applications for internet telephony solutions such as Skype have been available for some time. However there is still a higher degree of effort required by users to use these mobile internet telephony clients, as the user must first launch the application, before accessing the address book within the application to make outbound calls. Similarly for inbound calls, the user must already be visible as ‘logged in’ or ‘online’ in the internet telephony solution's presence status server before incoming calls can be accepted.
Another problem holding back the wider adoption of VoIP services in mobile communication networks such as 3G WCDMA is that the VoIP packet-based technology does not map well to the characteristics of cellular networks. In general terms, for a user to experience a good-quality natural conversation on a two-way voice call, it is necessary to ensure that the total delay in relaying the voice signal is kept low. If the end-to-end latencies are low, those engaged in the two-way conversation do not usually notice the delay in the other party's replies. If the delay times are long, there will appear to be unnaturally long silences before the other party replies. The delay experienced in each direction is a factor of the sample size of voice encoded, the encoding process, the transmission time, and the decoding and playback time. It is estimated that the round-trip delays for cellular transmission network should be less than 100 milliseconds and preferably significantly less (e.g. 50 milliseconds). Typical packet data latencies for Release 99 3GPP UMTS cellular networks are of the order of 200 milliseconds round-trip delay. With more recent releases of 3G such as the HSDPA and HSPA networks, the round-trip delay is being reduced towards the required target, but even then the bandwidth used for VoIP call is potentially large compared to that required by a circuit-switched radio bearer designed to accommodate a 12.2 kbps voice signal (e.g using GSM Full Rate or AMR codecs).
Given that packet-based VoIP services for 3G networks are still immature and inefficient, services such as Skype have also been implemented by way of gateways between circuit-switched and packet-switched networks. In the case of the mobile operator 3, it is already known that Skype calls are currently implemented as circuit-switched calls over the cellular network to a network-based gateway that then converts the signal to a packet-based Skype call.
Furthermore, the inventors of the present invention have identified a problem in that anyone who wishes to make use of multiple communication services over an IP network would currently be required to be very proactive to control the various client programs that control the various communication services. The inventors have identified a number of other problems that currently result in a very fragmented user experience.